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webrtc_m130/webrtc
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deadbeef 90f1e1e0d7 Fixing SDP parsing crash due to invalid port numbers. 
BUG=chromium:677029

Review-Url: https://codereview.webrtc.org/2675273003
Cr-Commit-Position: refs/heads/master@{#16541}
2017-02-10 20:35:05 +00:00
..
api
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
audio
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
base
Fixing SDP parsing crash due to invalid port numbers.
2017-02-10 20:35:05 +00:00
call
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Remove henrik.lundin from webrtc/common_video/OWNERS
2017-02-10 16:44:31 +00:00
examples
Move Android video quality loopback script.
2017-02-09 09:05:54 +00:00
logging
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
media
Rename "PacketTransportInterface" to "PacketTransportInternal".
2017-02-10 19:31:50 +00:00
modules
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
p2p
Rename "PacketTransportInterface" to "PacketTransportInternal".
2017-02-10 19:31:50 +00:00
pc
Fixing SDP parsing crash due to invalid port numbers.
2017-02-10 20:35:05 +00:00
sdk
Roll chromium_revision baaeb3f30e..ba3e8dd8fa (449272:449304)
2017-02-09 19:19:31 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
replace NtpTime->Clock with Clock->NtpTime dependency
2017-02-09 19:15:25 +00:00
test
Make functions in fileutils.h use "const std::string&".
2017-02-10 17:27:14 +00:00
tools
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
video
Retransmitted packets are now counted in receive time
2017-02-10 17:16:05 +00:00
voice_engine
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
.gitignore
…
BUILD.gn
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Check for use_x11 before runnig desktop_capture_modules_tests on linux.
2017-02-10 09:38:23 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.